I have been recording on a Tascam 388 for a while and I just bought an M-Track 8 to transfer to digital. I'm using garage band. I am very new to digital recording and I'd like some tips on handling the basics of levels.

There are three places to set your levels as far as I can tell, the individual 388 tape track RCA bus sends, the digital interface inputs, and computer record level. I want to make sure I'm getting the best possible sound without clipping.

What do you all recommend?

I tried having my 388 tracks at a healthy level, just kissing the red. I had the interface tracks set mostly in the green but occasionally tipping to the yellow, but that was like 2-3 o clock on the level knobs.

Going into garageband that was making a really healthy wave form that was still dynamic (not flat on the top and bottom).

One concern I had is the master fader at the top was totally running all the way into the red, and I had to bring the master fader down quite a bit to eliminate clipping, is that okay? Also it seems like I'm having a difficult time producing a mix down that is loud without clipping.

It seems like I should be judging based on the wave forms in garage band, correct?

1) make the 388 sound however you like it the best regardless of anything

2) properly connect the 388 to the ADC (interface) if it's all unbalanced connections use the shortest best quality cable if the interface is balanced input only you'll need to convert it with a transformer on each line for it to be optimal - all this not 100% necessary though it's the best way

3) determine where 0dB VU corresponds to on your interface metering if it has meters - if not use the software channels - (call the company or check the manual for this info - it's usually somewhere around -20dBFS to -12dBFS) - the goal is to adjust your interface input gain so that the tracks average and not peak at that spot on the meters - in modern 24 bit computer recording just using the halfway point on your meters will usually be just fine and shouldn't sound noticably different - just don't clip the converters (top of interface meter) and it it'll be acceptable

after you digitize if the tracks are summing where the master DAW is clipping then:

4) keep the master fader at unity/neutral/no gain boost as well as the channel faders - insert the trim/gain plug-in on the individual channels and reduce gain from there - mixing into the master where the sum of everything doesn't clip the output anymore

I tried having my 388 tracks at a healthy level, just kissing the red. I had the interface tracks set mostly in the green but occasionally tipping to the yellow, but that was like 2-3 o clock on the level knobs.

Going into garageband that was making a really healthy wave form that was still dynamic (not flat on the top and bottom).

One concern I had is the master fader at the top was totally running all the way into the red, and I had to bring the master fader down quite a bit to eliminate clipping, is that okay? Also it seems like I'm having a difficult time producing a mix down that is loud without clipping.

It seems like I should be judging based on the wave forms in garage band, correct?

Hi akpasta,

I used to do tape transfers for lots of people back in the day. Sometimes weird formats, for which the client would bring the deck to me.

Here are rules of thumb for tape transfers. The goal here is for them to be ACCURATE. You do not want to introduce ANY distortion ANYWHERE in the signal chain, or really, you will end up regretting it later on.

1. Set your tape output levels with a calibration tape, if you have one. Then, run the material once, checking for distortion (sometimes recordings went into the red, so a particular channel would be calibrated lower, this happened more often than not. the rare or ocassional peak is not as crucial as a track that is simply always loud.) Setting a uniform playback level is very important, and avoiding hotly recorded tracks is also important.

2. Calibrate your audio interface to be also uniform. You do this by playing back a 1kHz sine wave into each input, and setting it to one of these levels: -14dBFS, or -18dBFS, which will give you a standard level. These are very common level references in pro audio. You can play back a sine wave from your DAW, just mute the input channels as you do so, or send the channels not to your main output but a bus that goes nowhere.

3. Do not worry about getting the playback level on the digital interface to be louder, if they are not. Post processing gain can be applied easily with a plug in, avoiding distortion. You can always ADD gain, you cannot UNDO distortion.

4. From your description of what you have tried already, you are already making mistakes. Stop, slow down, and start over.

Important tip: Digital meters are also NOT accurate. They will not tell you if you are doing a type of distortion called "intersample peak", which means when a waveform peaks in between a sample being recorded. When it doubt, TURN IT DOWN.

1) make the 388 sound however you like it the best regardless of anything

2) properly connect the 388 to the ADC (interface) if it's all unbalanced connections use the shortest best quality cable if the interface is balanced input only you'll need to convert it with a transformer on each line for it to be optimal - all this not 100% necessary though it's the best way

3) determine where 0dB VU corresponds to on your interface metering if it has meters - if not use the software channels - (call the company or check the manual for this info - it's usually somewhere around -20dBFS to -12dBFS) - the goal is to adjust your interface input gain so that the tracks average and not peak at that spot on the meters - in modern 24 bit computer recording just using the halfway point on your meters will usually be just fine and shouldn't sound noticably different - just don't clip the converters (top of interface meter) and it it'll be acceptable

after you digitize if the tracks are summing where the master DAW is clipping then:

4) keep the master fader at unity/neutral/no gain boost as well as the channel faders - insert the trim/gain plug-in on the individual channels and reduce gain from there - mixing into the master where the sum of everything doesn't clip the output anymore

Hi "I'm Painting Again",

I have to disagree, with respect, on point 1 and point 4. When doing transfers to another format, the goal is not "to make it sound good", but accuracy. This means all levels have to be equal to each other, and referencing a known output level. Otherwise, all sorts of "fun" things can happen. Like a track having too much tape hiss noise, than during mixing a few weeks after the fact, you cannot take the hiss away. Percussions are normally recorded at a lower level on tape, to preserve their transient attack. I have seen a few times engineers crank those very percussion tracks, only to lament it afterwards.

On point 4, All you have to do is set the individual channel faders down by an equal amount for monitoring purposes. Introducing a plug in, especially in some consumer DAW like Garageband, can end up causing too much CPU usage which is totally unnecessary. The faders are already there for this purpose. We do not know how powerful his computer is, so the less strain on it, the better. I think this idea of "all faders at unity, and set the trim different" comes from a live engineering standpoint. It definitely is not from a studio standpoint. Lastly, having a plug in which trims down the levels you see at the meter inside the DAW, might make your forget that the level hitting the converters is louder. And you may miss it if it distorts. Again, the point here is accuracy, and it is harder with one more gain step in the chain.

1) make the 388 sound however you like it the best regardless of anything

2) properly connect the 388 to the ADC (interface) if it's all unbalanced connections use the shortest best quality cable if the interface is balanced input only you'll need to convert it with a transformer on each line for it to be optimal - all this not 100% necessary though it's the best way

3) determine where 0dB VU corresponds to on your interface metering if it has meters - if not use the software channels - (call the company or check the manual for this info - it's usually somewhere around -20dBFS to -12dBFS) - the goal is to adjust your interface input gain so that the tracks average and not peak at that spot on the meters - in modern 24 bit computer recording just using the halfway point on your meters will usually be just fine and shouldn't sound noticably different - just don't clip the converters (top of interface meter) and it it'll be acceptable

after you digitize if the tracks are summing where the master DAW is clipping then:

4) keep the master fader at unity/neutral/no gain boost as well as the channel faders - insert the trim/gain plug-in on the individual channels and reduce gain from there - mixing into the master where the sum of everything doesn't clip the output anymore

Hi "I'm Painting Again",

I have to disagree, with respect, on point 1 and point 4. When doing transfers to another format, the goal is not "to make it sound good", but accuracy. This means all levels have to be equal to each other, and referencing a known output level. Otherwise, all sorts of "fun" things can happen. Like a track having too much tape hiss noise, than during mixing a few weeks after the fact, you cannot take the hiss away. Percussions are normally recorded at a lower level on tape, to preserve their transient attack. I have seen a few times engineers crank those very percussion tracks, only to lament it afterwards.

On point 4, All you have to do is set the individual channel faders down by an equal amount for monitoring purposes. Introducing a plug in, especially in some consumer DAW like Garageband, can end up causing too much CPU usage which is totally unnecessary. The faders are already there for this purpose. We do not know how powerful his computer is, so the less strain on it, the better. I think this idea of "all faders at unity, and set the trim different" comes from a live engineering standpoint. It definitely is not from a studio standpoint. Lastly, having a plug in which trims down the levels you see at the meter inside the DAW, might make your forget that the level hitting the converters is louder. And you may miss it if it distorts. Again, the point here is accuracy, and it is harder with one more gain step in the chain.

Points 2 and 3 are very good!

Cheers!

yea accuracy is definitely the point of all artistic endeavors

My point one is simply get it the way they like coming off the tape machine - and if they want hiss that's their prerogative - mastering engineers routinely play with non optimal gain staging to make the masters of our classic records - they routinely CLIP converters in order to make it LOUD - like c'mon - most accurate possible is one path to choose sure but it aint for anyone to say it's better - just different - it's whatever the artist wants to do - and I'm pretty sure this is an artist asking not an aspiring duplication house proprietor

this is a transfer and mastering art question as well because of the fact that they stated things like "seems like I'm having a difficult time producing a mix down that is loud without clipping" etc. that particular statement also implies what they're doing is coming off the tape multichannel and mixing in the box as part of their process

As for four - I think it's pretty standard to use trim plug-ins to establish workable headroom in the master buss of a DAW - one main reason being DAW faders usually aren't symmetrical and you can fine tune them better the closer they are to unity - as to the horsepower of akpasta's computer - yea I have no idea if it can handle or even has trim plug-in - never used the software - but chances are it'll be the way to do it

It's funny you used the pronoun "him" to refer to akpasta - do you know akpasta personally?

Again If you read what the OP is telling us you'll see this isn't a flat transfer question - they seem to be multi outputting off the 388, having problems with the master buss clipping and overall apparent volume with the 8 summed tracks ITB - it's a hybrid recording and mastering question, no?

Am I reading this wrong?

We're not discussing your tape transfers here I'm sure they're great - we're discussing someone apart from us having problems with their music recording tech process

There's a time and place for everything and anything but don't front like what I'm saying here isn't normal

mastering engineers often clip ADC's (or software or analog outboard) to get a limiting effect that preserves the transient punch rather than using a limiter that will usually soften it / impart its particular unique artifacts - it's a good combination of the two in careful doses (or none of the above) if they're competent - yes it's distorted (sometimes unnoticeably - sometimes a whole lot!) - but it happens due to desires for loud punchy program material, etc. - it's a way to get a little more level to be competitive

I can clip my Lavry Blues much more than a prosumer ADC I worked with without noticeable distortion for example - this often is the clipping of the analog input before the ADC chip - each piece is different and what you can get away with varies accordingly

and again, DAW gain staging using trim to get your input signal level to hang around the fader's unity in order for them to be in a more workable range for mixing and to establish headroom on the mixbus seems pretty darn widely implemented in my experience

one can pull down DAW faders into the range where a tiny physical move equates to an enormous boost/cut if one desires tho - no rules ! - if it works it's good - if it sounds good it is good !

It sounds like the basic recommendation is to get a good healthy level on the analog machine, set the m-track 8 to "unity" or perhaps just 12 o'clock on each channel, and don't worry about the channel meters--which don't seem very accurate.

Then the computer side, do I want to be adjusting recording levels a lot there? Should I set each track to "0db"? Should I adjust the overall recording level from the Mac OS settings? I read elsewhere to leave the overall input recording level in the Mac OS settings to full.

A lot obviously depends on your specific set-up. I have an m-audio 1010lt I've used in the past for this sort of thing, which has a "hardware control" dialog window that allows each line input/output to be set as either -10 or +4 = 0dB. Then I can set up a test tone in the DAW and do a round trip level check, to confirm what levels correspond on different meters. For instance, what level out of the daw/ interface corresponds to 0dB on a mixer line in, tape input, etc, whether 0dB going to tape corresponds to 0dB coming from tape, and then what do I have to do to get that to come back out of the mixer and into the DAW at the same level. It's also useful to know whether, say, track 1 and track 4 are both operating at the same level when the entire chain is set identically, so that I can then better judge differences in program levels. Knowing this sort of stuff then makes it easier to systematically experiment with where to apply more or less gain - such as at the mix bus vs at the interface input vs in the dialog window- and determine if one method sounds better than the other (usually, which one distorts when).
You mention running the 388 "kissing the red," you might experiment with that during both tracking and transfer, especially on highly transient stuff like drums, to see if different levels change the sound in a useful way.

As for trimming levels once you're in the DAW (assuming it's got in just fine, I just don't need that hot a signal in my mix), this is probably the totally wrong way to do it but I sometimes just trim the volume and render a new track so as to save cpu and have faders near unity. But in general I find if i get individual track levels right to begin with - and that usually means lower levels - this sort of thing is less of an issue. 24 bit audio has plenty of headroom so why not use it.

On the initial recording in question I was able to set the master on my garage band at 0db without any clipping. I think I was confused about something. It seems my initial transfer was done "properly." But I'm noticing something new...

I have some other demos mixed from the 388 to garage band just via the main RCA L/R outs to a computer via RCA to headphone jack and they are much fuller and dynamic than the song I transferred via 8 track bus-out to the m-track 8 to garage band, did overdubs on and mixed down via garage band.

- It could either be RCA L/R Outs on the 388 versus the RCA bus outs for each track, maybe going through the master channel on the 388 adds some punch and color.

- It could be a difference of running straight to the headphone jack on a computer versus running through the M-Track 8 audio interface.

- It could be a difference of computers/versions of garage band. The fuller 2-track version was mixed to a VERY old macbook that is using old garage band that exports to mp3 at 320kbps. The new computer with the new garage band only allows a mp3 mix at 256kpbs. A funny little devolution, maybe meaningless, maybe not.

I need to experiment a little more to see exactly where the difference is. Does anyone have any ideas?

mastering engineers often clip ADC's (or software or analog outboard) to get a limiting effect that preserves the transient punch rather than using a limiter that will usually soften it / impart its particular unique artifacts - it's a good combination of the two in careful doses (or none of the above) if they're competent - yes it's distorted (sometimes unnoticeably - sometimes a whole lot!) - but it happens due to desires for loud punchy program material, etc. - it's a way to get a little more level to be competitive

Probably not any ME I'd want to use. But hey, whatever.

The one time an ME did this to my mixes, the client got super pissed off. They ended up using my mixes "sans mastering" because the goofball took too long, and was arrogant AF.

But again, if you want distortion, then do it however you want. I come from a different school I guess.

mastering engineers often clip ADC's (or software or analog outboard) to get a limiting effect that preserves the transient punch rather than using a limiter that will usually soften it / impart its particular unique artifacts - it's a good combination of the two in careful doses (or none of the above) if they're competent - yes it's distorted (sometimes unnoticeably - sometimes a whole lot!) - but it happens due to desires for loud punchy program material, etc. - it's a way to get a little more level to be competitive

Probably not any ME I'd want to use. But hey, whatever.

The one time an ME did this to my mixes, the client got super pissed off. They ended up using my mixes "sans mastering" because the goofball took too long, and was arrogant AF.

But again, if you want distortion, then do it however you want. I come from a different school I guess.

On the initial recording in question I was able to set the master on my garage band at 0db without any clipping. I think I was confused about something. It seems my initial transfer was done "properly." But I'm noticing something new...

I have some other demos mixed from the 388 to garage band just via the main RCA L/R outs to a computer via RCA to headphone jack and they are much fuller and dynamic than the song I transferred via 8 track bus-out to the m-track 8 to garage band, did overdubs on and mixed down via garage band.

- It could either be RCA L/R Outs on the 388 versus the RCA bus outs for each track, maybe going through the master channel on the 388 adds some punch and color.

- It could be a difference of running straight to the headphone jack on a computer versus running through the M-Track 8 audio interface.

- It could be a difference of computers/versions of garage band. The fuller 2-track version was mixed to a VERY old macbook that is using old garage band that exports to mp3 at 320kbps. The new computer with the new garage band only allows a mp3 mix at 256kpbs. A funny little devolution, maybe meaningless, maybe not.

I need to experiment a little more to see exactly where the difference is. Does anyone have any ideas?

it's probably not the software

hard to say what it could be - could be just the different material was recorded differently - ya know?

I do know that the 388 only outputs the EQ on certain outputs - that might be the problem ?

also headphone outs definitely sound different than line outs pretty much every time - I'd suggest just putting the same signal through each and comparing to see what exactly the differences are

and that's the thing..

to really get it the way you want you gotta spend time listening to every setting trying EVERYTHING - like no one can tell you a right way - especially with old analog gear - each unit sounds different and it's not about some "right way" but training your ears to compensate for what's going on in front of you in real time with the specific tools you have at in hand to achieve the results you want